1. Technical Field
The present invention relates to a voice over Internet protocol (VoIP) service and, more particularly, to a method and apparatus for signaling a VoIP call based on a class of service in a VoIP service system, the method and apparatus being capable of dynamically performing VoIP signaling based on the service class to set up a call by discriminating and setting various quality of service (QoS) factors based on the service class depending on a user policy or a VoIP operator's policy in the VoIP service system.
2. Related Art
Recently, a voice over Internet protocol (VoIP) technique that transfers voice over a packet network is attracting much attention. VoIP-based Internet phone service has been put to practical use since 1995 and, in Korea, various services have been provided by lawful Internet phone providers since January, 1998 under the revised telecommunication business law, allowing significantly inexpensive international calls compared to existing telephone charges. In addition, a free service which was begun in autumn 1999 provided a chance to attract the attention of the public with respect to Internet phone service. Existing telephone network providers are now experiencing a crisis, and the telephony service market has been said to be in need of reform.
At present, a representative example of the VoIP technique includes the IP telephony service which is mainly provided over an Intranet network using an IP-PBX.
Free VoIP-based Internet telephony service, which has recently achieved much popularity due to its openness, spreads recognition on the Internet phone to public users, and causes significant change in the national economic planning led by large-scaled enterprise groups. Furthermore, with Is the evolution of Internet network capability and associated techniques, it is predicted that, in the ultimate, all communication systems and the Internet will be incorporated, or Internet based communications will be substituted for existing communications.
The voice over Internet protocol (VOIP) is a technology that transfers voice information having continuity and real-time features over an Internet characterized by a packet switched network. The voice over Internet protocol is applicable to various application fields but is being currently used with substantially the same meaning as the Internet telephony. Friendliness of the Internet telephony to the public is because of its lower communication cost. Even though free PC-to-telephony service has not begun, an inexpensive PC-to-PC Internet phone, inexpensive international telephony by a voice resale provider, and the like have already attracted attention. Initially, existing telephony providers, who feel concern about rapid market encroachment, have even raised a lawful issue.
The VoIP-based Internet phone service has competitive pricing for two reasons: first, the Internet phone utilizes an Internet backbone network which is already disposed and utilized as a data transmission network; and second, the Internet phone uses a packet transmission system, and thus it is able to use the network more efficiently as compared to the existing telephone network, which exclusively occupies a line in a busy state, thus reducing a communication cost.
However, this may be a cause of degrading capability of the Internet phone, and thus various techniques for overcoming this problem are required. An alternative has been suggested, and it calls for real-time data (e.g., voice and image) and a data transmission network which should be separated virtually or substantially.
In addition to competitive pricing, the VoIP-based Internet phone has the advantages of flexible utilization of bandwidth, ease of differentiated services, incorporation with various services, and the like. The VoIP Internet phone has further advantages in that the VoIP phone service costs the same irrespective of calling distance, and the VoIP phone is easy to access because the VoIP phone is available wherever the Internet is disposed.
On the other hand, the Internet phone has many technical problems to be solved. Retardation and unsatisfactory communication quality caused by the capability of networks and terminals, interoperability between various device manufacturers that use their own way, security, the provision and operation of various services available for existing telephones, the creation of new services to effectively combine other functionalities of the Internet, and the like are imminent problems.
The evolution and generalization of the VoIP may be a way in which these problems get successfully solved. Some outcomes obtained by various efforts in the art have yielded the current services.
The VoIP, which is a service for conveying image, voice and facsimile messages over the Internet, transmits real-time media such as voice and video, for example, where a user desiring to use the Internet accesses the Internet by means of his or her PC, by means of an independent device operating with an Internet protocol, or by calling to a gateway at an existing public switched telephone network (PSTN) terminal.
With the VoIP service, there is a need to discover and signal a correspondent. Examples of the VoIP signaling include H.323 of the International Telecommunication Standardization Sector (ITU-T), and session initiation protocol (SIP) of the Internet Engineering Task Force (IETF).
In a conventional VoIP service system, key factors for the VoIP service, such as the type of VoIP signaling (e.g., H.323, SIP, MGCP, MEGACO, and H.248), the type of used codec, silence suppression or non-silence suppression, and a count of multiframes, have been set as fixed values for the sake of the VoIP signaling.
Thus, the conventional VoIP service system does not employ a class of service (CoS) concept in which services are classified and provided by the service class, but rather employs a CoS in which the same quality of service (QoS) is applied to all users.
For this reason, an ISP or private network which provides the VoIP service uses a codec having a short sampling period so as to guarantee some VoIP QoS when performing the VoIP service. The reason of using a codec having a short sampling period relates to network bandwidth.
That is, since the G.711 codec uses a bandwidth exceeding the bandwidth per call (64 kbps) used in a public switched telephone network (PSTN), it is difficult to expect efficient bandwidth use in the performance of voice data compression for VoIP service. This is a cost burden for the VoIP provider who leases and uses a line.
Thus, the prior art has difficulty in providing the VoIP CoS because the VoIP terminal or VoIP server for the VoIP service applies the same VoIP QoS factor to all calls. That is, the same signaling capability negotiation is applied when handling all VoIP calls.
Accordingly, a codec having good mean opinion score (MoS) and perceptual evaluation of speech quality (PESQ) is used to guarantee QoS to all users, which causes a problem of overuse of the frequency band.
Meanwhile, there is no great difference in conscious voice sensitivity between codecs when a stabilized network, like a dedicated line, is used. However, when streaming data is VoIP-processed (e.g., color ring for a cell phone) like music data, there may be a difference in the conscious sensitivity between the codecs.
Furthermore, in a VoIP network where common data and voice are used together, there may be a significant difference in voice quality depending on VoIP media factors (e.g., codec, silence suppression, and multiframe count).
For this reason, the VoIP service provider uses a G.711 codec, which is used in the PSTN having a short sampling period, and which causes the overuse of the frequency band and thus waste of a resource.